commit fcc398163304ba0c26df5ff270905e349d3accc3
author Ilya Nikolaevskiy <ilnik@webrtc.org> Tue Oct 30 15:47:48 2018
committer Commit Bot <commit-bot@chromium.org> Tue Oct 30 15:47:59 2018
tree b3eb9c573cc31aecf9b390267bd2ad980d37dd93
parent 992a868393f322593e69d8a44d79dcb8c2f32a46

Revert "Use only first payload timestamp for RTCP SR generation for audio"

This reverts commit 9a0662ac7e4a3bc6b3a316397a7fdf25f0025d35.

Reason for revert: breaks some av sync perf tests

Original change's description:
> Use only first payload timestamp for RTCP SR generation for audio
> 
> Since now RTP rate is set correctly for audio, there's no need to
> use the very last data packet rtp/capture timestamps for generating
> RTCP SR packets.
> 
> Using only one (first) packet timestamp eliminates the jitter between
> rtp and capture timestamps for audio. This jitter comes from the fact
> that capture timestamp for audio is unknown and we generate bogus
> timestamp at arbitrary, non-constant offset from the real capture time.
> 
> Bug: webrtc:9905
> Change-Id: I855556184cfe994be39ab7780836a050f5a38c35
> Reviewed-on: https://webrtc-review.googlesource.com/c/108580
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25430}

TBR=danilchap@webrtc.org,ilnik@webrtc.org,ossu@webrtc.org

Change-Id: I208a659379b1075258ee94613e42afd9aebe4754
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9905
Reviewed-on: https://webrtc-review.googlesource.com/c/108623
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25435}