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platform-external-webrtc/webrtc
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danilchap 70f39a30e9 In RtpPacket do not keep pointer to RtpHeaderExtensionMap 
Having that dependency require user of RtpPacket to ensure
RtpHeaderExtensionMap always outlive packet and that RtpPacket's access
to RtpHeaderExtensionMap is properly syncrhonized.
Dropping this dependencies make use of RtpPacket less error-prone.

BUG=webrtc:5261

Review-Url: https://codereview.webrtc.org/2576653003
Cr-Commit-Position: refs/heads/master@{#15653}
2016-12-16 13:48:18 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
2016-12-15 14:17:15 +00:00
audio
Add ossu@ to OWNERS of audio/ and modules/audio_coding/
2016-12-15 15:52:14 +00:00
base
Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2581663002/ )
2016-12-15 11:54:52 +00:00
build
MB: Enable memcheck for the linux_memcheck trybot.
2016-12-15 10:09:17 +00:00
call
Disable failing perf test on Android.
2016-12-09 14:15:22 +00:00
common_audio
Fix for left shift of negative value in NetEq.
2016-12-14 10:00:02 +00:00
common_video
Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
2016-12-15 14:30:00 +00:00
examples
iOS: Add trendline filter to field trials.
2016-12-15 21:17:58 +00:00
libjingle
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
logging
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
media
Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
2016-12-15 14:30:00 +00:00
modules
In RtpPacket do not keep pointer to RtpHeaderExtensionMap
2016-12-16 13:48:18 +00:00
p2p
Fixing possible crash due to RefCountedChannel assignment operator.
2016-12-14 00:38:46 +00:00
pc
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
sdk
Move ios_helpers to sdk folder
2016-12-16 10:12:42 +00:00
stats
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
2016-12-15 14:17:15 +00:00
system_wrappers
Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002
2016-12-14 08:07:44 +00:00
test
Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
2016-12-15 14:30:00 +00:00
tools
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
video
Fix wrong log message.
2016-12-16 11:46:48 +00:00
voice_engine
Wire-up audio BWE with overhead.
2016-12-15 14:24:52 +00:00
.gitignore
…
BUILD.gn
Adds basic Bluetooth support to AppRTCMobile
2016-12-14 15:37:01 +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
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
2016-12-09 10:35:30 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +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
…
README.chromium
…
typedefs.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
2016-12-15 14:30:00 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +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.