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zhihuang f816493c4f Add media related stats (audio level etc.) to unsignaled streams. 
The media related stats wasn't working for unsignaled stream because there
is no mapping between the receiver_info and unsignaled tracks.

This CL fixes the issue by adding some special logic to the TrackMediaInfoMap
which would create the mapping.

BUG=b/37836881
BUG=webrtc:7685

TBR=deadbeef@webrtc.org

Review-Url: https://codereview.webrtc.org/2883943003
Cr-Commit-Position: refs/heads/master@{#18217}
2017-05-19 20:09:47 +00:00
..
api
Renaming probing_interval to bwe_period globally.
2017-05-18 21:32:41 +00:00
audio
Renaming probing_interval to bwe_period globally.
2017-05-18 21:32:41 +00:00
base
Land ContextUtils separately.
2017-05-19 08:29:10 +00:00
call
New class RtxReceiveStream.
2017-05-19 13:15:19 +00:00
common_audio
Moving compile_assert_c.h to webrtc/base
2017-05-15 15:06:47 +00:00
common_video
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
examples
AppRTCMobile support to turn off the WebRTC AGC and HPF
2017-05-18 22:09:33 +00:00
logging
Change rtp_event_log2text to ignore webrtc::MediaType from proto.
2017-05-19 12:32:56 +00:00
media
Reduce VideoSendStream recreations due to FlexFEC.
2017-05-19 12:47:46 +00:00
modules
Revert of Reuse allocated encoders in SimulcastEncoderAdapter. (patchset #15 id:320001 of https://codereview.webrtc.org/2830793005/ )
2017-05-19 17:33:03 +00:00
ortc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
p2p
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
pc
Add media related stats (audio level etc.) to unsignaled streams.
2017-05-19 20:09:47 +00:00
sdk
Fixing memory leak of generated session descriptions on Android.
2017-05-18 15:45:59 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Adding target to track asm_defines.h
2017-05-15 21:59:17 +00:00
test
Renaming probing_interval to bwe_period globally.
2017-05-18 21:32:41 +00:00
tools
Change how event_log_visualizer ignore duplicate incoming RTCP packets.
2017-05-19 13:30:28 +00:00
video
Revert of Reuse allocated encoders in SimulcastEncoderAdapter. (patchset #15 id:320001 of https://codereview.webrtc.org/2830793005/ )
2017-05-19 17:33:03 +00:00
voice_engine
Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()
2017-05-03 15:30:15 +00:00
.gitignore
…
BUILD.gn
Revert of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2890733003/ )
2017-05-17 13:52:58 +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
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +00:00
config.cc
Fix default extension id for video content type extension
2017-04-26 09:35:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_frame.h
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +00:00
webrtc.gni
Replace sanitizers:deps with exe_and_shlib_deps (webrtc)
2017-05-19 06:51:46 +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.