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sakal 1fd9595936 Pass VideoDecoderParams to VideoDecoderFactory and add SSRC to RtpEncodingParameters. 
VideoDecoderParams contains the id of the receive video
stream. Motivation behind this change is to enable down
stream apps easier map raw non-decoded data to incoming
streams.

BUG=b/28636393

Review-Url: https://codereview.webrtc.org/2052233002
Cr-Commit-Position: refs/heads/master@{#13250}
2016-06-22 07:46:19 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Pass VideoDecoderParams to VideoDecoderFactory and add SSRC to RtpEncodingParameters.
2016-06-22 07:46:19 +00:00
audio
Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ )
2016-06-20 18:53:09 +00:00
base
Use the new API to set the BoringSSL time callback.
2016-06-20 19:00:48 +00:00
build
Remove dead GYP target audio_device_module_java
2016-06-20 08:43:42 +00:00
call
Remove ViEEncoder::Pause / Start
2016-06-17 14:27:23 +00:00
common_audio
GN: Add common_audio_unittests and common_video_unittests
2016-06-08 08:28:15 +00:00
common_video
Delete deprecated VideoFrameBuffer methods.
2016-06-21 10:59:32 +00:00
examples
Add RTCEventLog API to ObjC.
2016-06-14 22:03:19 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Pass VideoDecoderParams to VideoDecoderFactory and add SSRC to RtpEncodingParameters.
2016-06-22 07:46:19 +00:00
modules
Remove DesktopCapturer::Callback::OnCaptureCompleted()
2016-06-21 23:50:07 +00:00
p2p
Update ICE role on all ports, not just ones used for new connections.
2016-06-21 21:19:55 +00:00
pc
Delete GetExecutablePath and related unused code.
2016-06-16 10:44:44 +00:00
sdk
Revert of Delete method cricket::VideoFrame::Copy. (patchset #7 id:120001 of https://codereview.webrtc.org/2080253002/ )
2016-06-21 21:03:01 +00:00
system_wrappers
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
test
Use the Chromium libfuzzer template instead of rolling our own.
2016-06-20 13:04:01 +00:00
tools
New rtc dump analyzing tool in Python
2016-06-20 14:39:21 +00:00
video
Reland of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #1 id:1 of https://codereview.webrtc.org/2084873002/ )
2016-06-21 07:26:48 +00:00
voice_engine
Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ )
2016-06-20 18:53:09 +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
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
GN: Add video_engine_tests
2016-06-14 10:53:09 +00:00
call.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +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
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Partial reland of Delete unused and almost unused frame-related methods. (patchset #1 id:1 of https://codereview.webrtc.org/2076113002/ )
2016-06-17 12:03:09 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Workaround java.gypi inclusion error in Chromium builds.
2016-06-20 08:34:11 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 07:39:49 +00:00
webrtc_tests.gypi
GN: Add SDK tests to rtc_unittests.
2016-06-08 08:14:22 +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.