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henrika bb6f7524ba Improved audio buffer handling for iOS. 
This change:

Reduces complexity for audio playout by removing a redundant memcopy in the output audio path.

Adds support for iOS simulator for playout since we now allow the audio layer to ask for different sizes
of audio buffers at each callback. Real iOS devices always asks for the same size, simulators does not.
This change comes without any new cost for real devices.

BUG=b/37580746

Review-Url: https://codereview.webrtc.org/2894873002
Cr-Commit-Position: refs/heads/master@{#18321}
2017-05-30 09:01:30 +00:00
..
api
Add separate base classes for I420 and I444 buffers
2017-05-30 08:21:59 +00:00
audio
Rename elad.alon to eladalon, to avoid confusion between repositories.
2017-05-25 07:15:35 +00:00
base
Disable SIGPIPE for sockets created on iOS.
2017-05-26 23:27:09 +00:00
call
Store/restore RTP state for audio streams with same SSRC within a call
2017-05-23 13:07:11 +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
Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2911053002/ )
2017-05-30 08:48:47 +00:00
logging
Remove final from RtcEventLogNullImpl
2017-05-29 09:46:05 +00:00
media
Recreate FlexfecReceiveStream separately from VideoReceiveStream.
2017-05-30 08:31:37 +00:00
modules
Improved audio buffer handling for iOS.
2017-05-30 09:01:30 +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
Reland of Adding backward compatibility header (patchset #1 id:1 of https://codereview.webrtc.org/2908143003/ )
2017-05-30 08:50:35 +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
Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2911053002/ )
2017-05-30 08:48:47 +00:00
tools
Revert of Remove unecessary non fatal error statement that very often is printed in the PSNR or SSIM metric n… (patchset #1 id:1 of https://codereview.webrtc.org/2901793002/ )
2017-05-23 15:31:11 +00:00
video
Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead.
2017-05-29 15:16:37 +00:00
voice_engine
Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead.
2017-05-29 15:16:37 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2911053002/ )
2017-05-30 08:48:47 +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
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.