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Issue 3333 link

Starred by 18 users

Issue metadata

Status: Duplicate
Merged: issue chromium:703122
Owner: ----
Closed: Nov 2
Cc:
Components:
NextAction: ----
OS: ----
Pri: 3
Type: Bug



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CSRCs audio activity/contribution through getStats or other WebRTC API

Reported by os...@tokbox.com, May 12 2014

Issue description

What steps will reproduce the problem?
In many situations, mixed audio data is necessary in order to achieve efficient transport of audio in multi-party conferences. Audio is usually mixed on a MCU-like server and sent to every subscribing client. Even if audio is mixed, it may be needed to have the possibility to identify which audio source is contributing to the mixed audio at every moment at the subscribing client side.

This can be done in a basic manner indicating CSRCs on RTP packets (e.g. as recommended in 7.1 section of RFC3550), or in a more advanced manner through RFC 6465 spec (enabled by  urn:ietf:params:rtp-hdrext:csrc-audio-level on RFC).

What is the expected result?
Support of CSRC contribution display through getStats or other alternative through the WebRTC API.

What do you see instead?
It seems not supported

What version of the product are you using? On what operating system?
 34.0.1847.131 on Mac OS X

Any plans for supporting this? Timings?



 
Project Member

Comment 1 by braveyao@webrtc.org, May 14 2014

Cc: braveyao@webrtc.org vikasmarwaha@webrtc.org
Owner: juberti@webrtc.org
@justin, any comment?
This will not be supported in WebRTC 1.0. It is currently being debated for future versions of WebRTC.
Project Member

Comment 3 by juberti@webrtc.org, May 22 2014

Labels: Area-PeerConnection
Status: WontFix
Reopen if/when this lands in the W3C spec. Until then, you can use a datachannel from the mixer to the client.
https://github.com/w3c/webrtc-pc/pull/300 has been merged to 1.0 so this should be reopened.
Project Member

Comment 5 by juberti@webrtc.org, Oct 12 2015

Owner: pthatcher@webrtc.org
Status: Available
Indeed.
Project Member

Comment 6 by honghaiz@google.com, Oct 19 2015

Labels: EngTriaged
Project Member

Comment 7 by pthatcher@webrtc.org, Oct 26 2015

Yes, we'll add support for this as we add RtpSender and RtpReceivers according to the 1.0 spec.
Project Member

Comment 8 by pthatcher@webrtc.org, Nov 8 2016

Labels: Pri-3
Project Member

Comment 9 by anatolid@webrtc.org, Dec 14 2016

Cc: pthatcher@webrtc.org
Owner: ----
Project Member

Comment 10 by sheriffbot@chromium.org, Sep 6

Labels: Hotlist-Recharge-Cold
Status: Untriaged (was: Available)
This issue has been Available for over a year. If it's no longer important or seems unlikely to be fixed, please consider closing it out. If it is important, please re-triage the issue.

Sorry for the inconvenience if the bug really should have been left as Available.


Project Member

Comment 11 by anatolid@webrtc.org, Nov 1

Status: Available (was: Untriaged)
[bulk-edit] Setting status to Available since it's likely that this issue shouldn't be archived yet. Also changing Pri to 3 due to long period of inactivity (indicating low priority).
Project Member

Comment 12 by hbos@webrtc.org, Nov 2

Mergedinto: chromium:703122
Status: Duplicate (was: Available)
We have getSynchronizationSources and getContributingSources in the spec for this now. getCSRCs have landed, getSSRCs is WIP
Project Member

Comment 13 by bugdroid1@chromium.org, Nov 6

The following revision refers to this bug:
  https://webrtc.googlesource.com/src.git/+/967f7d549728f2c25b5d6f0ce2bab42478cf356e

commit 967f7d549728f2c25b5d6f0ce2bab42478cf356e
Author: Jonas Oreland <jonaso@webrtc.org>
Date: Tue Nov 06 12:10:05 2018

Add audio level to CSRC class

This patch adds (optional) csrc to ContributingSources.
This will be used if using virtual audio ssrc, since
the audio level is otherwise unaccessible in that configuration.

BUG= webrtc:3333 

Change-Id: Ied263b8f0850553cd637fd6bead373ed4252fd1e
Reviewed-on: https://webrtc-review.googlesource.com/c/109281
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: ├ůsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25516}
[modify] https://crrev.com/967f7d549728f2c25b5d6f0ce2bab42478cf356e/audio/channel_receive.cc
[modify] https://crrev.com/967f7d549728f2c25b5d6f0ce2bab42478cf356e/modules/rtp_rtcp/source/contributing_sources.cc
[modify] https://crrev.com/967f7d549728f2c25b5d6f0ce2bab42478cf356e/modules/rtp_rtcp/source/contributing_sources.h
[modify] https://crrev.com/967f7d549728f2c25b5d6f0ce2bab42478cf356e/modules/rtp_rtcp/source/contributing_sources_unittest.cc
[modify] https://crrev.com/967f7d549728f2c25b5d6f0ce2bab42478cf356e/video/rtp_video_stream_receiver.cc

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