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platform-external-webrtc/webrtc
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philipel 17deeb47ed PacketBuffer is now ref counted. 
Since all FrameObjects have a reference to its PacketBuffer and since
the PacketBuffer can be thrown away at any moment the PacketBuffer
has to be ref counted in order to avoid FrameObjects dereferencing a potentially
destroyed object.

BUG=webrtc:5514
R=danilchap@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2199133004 .

Cr-Commit-Position: refs/heads/master@{#13725}
2016-08-11 13:09:40 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Session based capturing for Camera2Capturer.
2016-08-11 08:00:43 +00:00
audio
Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.
2016-08-04 12:28:28 +00:00
base
Session based capturing for Camera2Capturer.
2016-08-11 08:00:43 +00:00
build
Add Mac bots to MB.
2016-08-11 08:04:39 +00:00
call
Generate random rtp packets with RtpPacketToSend instead of RtpSender
2016-08-10 11:23:38 +00:00
common_audio
Pull out the PostFilter to its own NonlinearBeamformer API
2016-07-02 00:19:32 +00:00
common_video
Reland of Switch to use SequencedTaskChecker instead of ThreadChecker where needed.
2016-07-15 06:36:00 +00:00
examples
Remove LooperExecutor and replace it with built-in HandlerThread.
2016-08-09 09:03:41 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
modules
PacketBuffer is now ref counted.
2016-08-11 13:09:40 +00:00
p2p
Do not create incompatible TurnPort if the server address family is known.
2016-08-10 17:33:17 +00:00
pc
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
sdk
iOS render: Handle frame rotation in OpenGL
2016-08-10 14:58:35 +00:00
system_wrappers
UMA log for audio_device Init and Start(Playout|Recording). Make Init return a more specific error code, if possible.
2016-07-01 11:35:31 +00:00
test
Style updates to ProducerFec/FecReceiver.
2016-08-10 07:51:59 +00:00
tools
Fix tools_unittests in GN.
2016-08-10 14:47:20 +00:00
video
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
voice_engine
Broke out 'level_indicator' of the voice_engine GN target. This is
2016-08-10 12:01:59 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Variable audio bitrate.
2016-07-26 11:44:12 +00:00
audio_sink.h
…
audio_state.h
…
BUILD.gn
GN: Add targets for AppRTCDemo on Mac+iOS
2016-08-08 04:37:56 +00:00
call.h
Fix bug where min transmit bitrate wasn't working
2016-07-06 07:54:35 +00:00
codereview.settings
…
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
2016-07-13 16:11:38 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add stefan@ to webrtc/OWNERS.
2016-08-10 15:49:59 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
…
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
…
video_engine_tests.isolate
…
video_frame.h
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
webrtc_nonparallel_tests_apk.isolate
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 07:39:49 +00:00
webrtc_tests.gypi
Reland Enable socketservers unittests on windows.
2016-08-05 12:55:49 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.