commit 67008dfb366237469fe088a61b62c0cad852c024
author Artem Titov <titovartem@webrtc.org> Thu Jul 04 07:14:11 2019
committer Commit Bot <commit-bot@chromium.org> Thu Jul 04 07:14:24 2019
tree c95835035b2a0c824f8e15a31a1cc5cad73b9955
parent c6c730b7fde0dada347d09edaddb77b4a3bba171

Revert "Replace the implementation of `GetContributingSources()` on the audio side."

This reverts commit 8fa7151e4bbad40fec1f964fe0c003b8787bb78a.

Reason for revert: Speculative revert to fix roll of webrtc into chrome. Right now tests related to RTCRtpReceiver failing and looks like it is main candidate, who can affect that behavior.

Original change's description:
> Replace the implementation of `GetContributingSources()` on the audio side.
> 
> This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the audio side with the spec-compliant `SourceTracker`-implementation.
> 
> The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network.
> 
> This change is almost identical to the previous video side change at: https://webrtc-review.googlesource.com/c/src/+/143177
> 
> Bug: webrtc:10545
> Change-Id: Ife7f08ee8ca1346099b7466837a3756947085fc5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144422
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Chen Xing <chxg@google.com>
> Cr-Commit-Position: refs/heads/master@{#28459}

TBR=ossu@webrtc.org,chxg@google.com

Change-Id: I5c631d4dcfb39601055ffce9b104f45eea871fd3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10545
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144562
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28478}