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mzanaty 8a855d6916 Allow any unsignalled SSRC changes on default video receive channel. 
The first unsignalled SSRC creates a default receive channel.
Any unsignalled SSRC changes after that replace the default SSRC.
Add unit tests for changing unsignalled SSRCs.

BUG=webrtc:5208

Review-Url: https://codereview.webrtc.org/2692993009
Cr-Commit-Position: refs/heads/master@{#16682}
2017-02-17 23:46:43 +00:00
..
api
Adding RTCErrorOr class to be used by ORTC APIs.
2017-02-17 07:31:33 +00:00
audio
Remove VoEVideoSync interface.
2017-02-15 08:42:31 +00:00
base
Use std::unique_ptr instead of rtc::scoped_refptr in AsyncInvoker.
2017-02-17 23:19:19 +00:00
call
Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.
2017-02-16 14:52:32 +00:00
common_audio
Fix for left shift of potentially negative value.
2017-02-09 11:05:59 +00:00
common_video
Added Vp9 simulcast tests.
2017-02-15 10:23:28 +00:00
examples
Modify android video_quality_loopback_test to run commands from the src dir.
2017-02-16 15:20:26 +00:00
logging
Delete class SSRCDatabase, and its global ssrc registry,
2017-02-17 16:34:35 +00:00
media
Allow any unsignalled SSRC changes on default video receive channel.
2017-02-17 23:46:43 +00:00
modules
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +00:00
p2p
Use rtc::ToString instead of std::to_string.
2017-02-16 19:10:51 +00:00
pc
Adding RTCErrorOr class to be used by ORTC APIs.
2017-02-17 07:31:33 +00:00
sdk
Relanding: Add the url attribute to the IceCandidate (Java Wrapper)
2017-02-17 20:45:00 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
Change NtpTime representation to single uint64_t
2017-02-15 09:18:15 +00:00
test
Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.
2017-02-16 14:52:32 +00:00
tools
Propagate packet pacing information to SenTimeHistory.
2017-02-17 11:59:43 +00:00
video
Replace DCHECK with ASSERT_TRUE in vie_encoder_unittest.cc
2017-02-16 23:46:27 +00:00
voice_engine
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +00:00
.gitignore
…
BUILD.gn
Adding RTCErrorOr class to be used by ORTC APIs.
2017-02-17 07:31:33 +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
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +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
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Move iOS tests to XCTest from gtest.
2017-02-14 12:58:56 +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.