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Peter Boström 3f4eed0cc0 Deliver RTCP packets only once per receive stream. 
Receive streams can have multiple entries in the SSRC table when RTX is
configured. This does not mean that they should receive RTCP packets
more than once.

BUG=4546
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43179004

Cr-Commit-Position: refs/heads/master@{#9019}
2015-04-16 13:59:35 +00:00
..
base
Cleanup: Remove unnecessary SHA1Transform() declaration.
2015-04-09 13:48:00 +00:00
build
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
common_audio
Fix unknown option '-msse2' warning
2015-04-16 06:57:12 +00:00
common_video
Remove i420_video_frame.h from common_video.gyp
2015-04-10 12:40:21 +00:00
examples/android
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
libjingle
Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
2015-04-04 23:56:56 +00:00
modules
Fix unknown option '-msse2' warning
2015-04-16 06:57:12 +00:00
overrides
Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
2015-04-04 23:56:56 +00:00
p2p
Remove PORTALLOCATOR_ENABLE_BUNDLE, PortAllocatorSessionProxy, PortAllocatorSessionMuxer, and PortProxy.
2015-04-14 16:25:58 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Remove android_webview_build conditions.
2015-04-09 15:36:13 +00:00
test
Fix for sent/received RTCP packet counters returned by GetRtcpPacketTypeCounters. The returned counters are incorrect: sent_packets returns stats from a sent stream (and received_packets returns stats from a receive stream).
2015-04-15 16:00:37 +00:00
tools
VoE: move mock directory 1 level up
2015-04-15 10:45:09 +00:00
video
Deliver RTCP packets only once per receive stream.
2015-04-16 13:59:35 +00:00
video_engine
Register RTP/RTCP modules outside rtp_rtcp_cs_.
2015-04-16 10:00:12 +00:00
voice_engine
VoE: move mock directory 1 level up
2015-04-15 10:45:09 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Let Chromium declare the mips_dsp_rev build variable.
2015-03-04 09:51:17 +00:00
call.h
Remove ViE interface usage from VideoReceiveStream.
2015-04-09 12:35:46 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
config.h
Add CVO support to Vie layer.
2015-03-12 20:51:50 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
2015-03-16 13:48:18 +00:00
video_encoder.h
Remove default arguments in EncodedImageCallback.
2015-02-09 09:14:48 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
2015-04-09 11:44:19 +00:00
video_receive_stream.h
Add decoder-timing stats to VideoReceiveStream.
2015-02-25 10:42:45 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
2015-03-18 09:51:44 +00:00
webrtc_examples.gyp
Remove video from WebRTC Android example.
2015-03-25 13:32:30 +00:00
webrtc_perf_tests.isolate
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
rtc_unittests on Android
2015-03-04 08:48:17 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.