Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(120)

Issues Search
    My Issues | Starred     Open | Closed | All

Issue 2667163002: Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (Closed)

Created:
3 years, 10 months ago by guidou
Modified:
3 years, 10 months ago
CC:
webrtc-reviews_webrtc.org, video-team_agora.io, yujie_mao (webrtc), zhengzhonghou_agora.io, stefan-webrtc, tlegrand-webrtc, tterriberry_mozilla.com, the sun, mflodman, philipel
Target Ref:
refs/heads/master
Project:
webrtc
Visibility:
Public.

Description

Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) Reason for revert: Suspect of breaking FYI bots. See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win7%20Tester/builds/7582 https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Tester/builds/34280 https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester Original issue's description: > Always call RemoteBitrateEstimator::IncomingPacket from Call. > > Delete the calls from RtpStreamReceiver (for video) and > AudioReceiveStream. > > BUG=webrtc:6847 > > Review-Url: https://codereview.webrtc.org/2659563002 > Cr-Commit-Position: refs/heads/master@{#16393} > Committed: https://chromium.googlesource.com/external/webrtc/+/6d4dd593a824cfc9c67f6e9fd3a08014d08350a0 TBR=stefan@webrtc.org,brandtr@webrtc.org,nisse@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6847

Patch Set 1 #

Unified diffs Side-by-side diffs Delta from patch set Stats (+63 lines, -77 lines) Patch
M webrtc/audio/audio_receive_stream.cc View 1 chunk +13 lines, -0 lines 0 comments Download
M webrtc/audio/audio_receive_stream_unittest.cc View 2 chunks +16 lines, -0 lines 0 comments Download
M webrtc/call/call.cc View 16 chunks +31 lines, -77 lines 0 comments Download
M webrtc/video/rtp_stream_receiver.cc View 2 chunks +3 lines, -0 lines 0 comments Download

Messages

Total messages: 7 (3 generated)
guidou
Created Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call.
3 years, 10 months ago (2017-02-01 17:19:29 UTC) #2
commit-bot: I haz the power
CQ is trying da patch. Follow status at https://chromium-cq-status.appspot.com/v2/patch-status/codereview.webrtc.org/2667163002/1
3 years, 10 months ago (2017-02-01 17:19:41 UTC) #3
commit-bot: I haz the power
Failed to apply patch for webrtc/audio/audio_receive_stream.cc: While running git apply --index -p1; error: patch failed: ...
3 years, 10 months ago (2017-02-01 17:19:48 UTC) #5
nisse-webrtc
3 years, 10 months ago (2017-02-02 07:48:43 UTC) #6 
On 2017/02/01 17:19:48, commit-bot: I haz the power wrote:
> Failed to apply patch for webrtc/audio/audio_receive_stream.cc:
> While running git apply --index -p1;
>   error: patch failed: webrtc/audio/audio_receive_stream.cc:330
>   error: webrtc/audio/audio_receive_stream.cc: patch does not apply
> 
> Patch:       webrtc/audio/audio_receive_stream.cc
> Index: webrtc/audio/audio_receive_stream.cc
> diff --git a/webrtc/audio/audio_receive_stream.cc
> b/webrtc/audio/audio_receive_stream.cc
> index
>
17da10f35789eb4c864ca6f696d3cfd09da91e9e..1f24b2ca2726c099cef35f215457d08425e6e19a
> 100644
> --- a/webrtc/audio/audio_receive_stream.cc
> +++ b/webrtc/audio/audio_receive_stream.cc
> @@ -330,6 +330,19 @@
>      return false;
>    }
>  
> +  // Only forward if the parsed header has one of the headers necessary for
> +  // bandwidth estimation. RTP timestamps has different rates for audio and
> +  // video and shouldn't be mixed.
> +  if (config_.rtp.transport_cc &&
> +      header.extension.hasTransportSequenceNumber) {
> +    int64_t arrival_time_ms = rtc::TimeMillis();
> +    if (packet_time.timestamp >= 0)
> +      arrival_time_ms = (packet_time.timestamp + 500) / 1000;
> +    size_t payload_size = length - header.headerLength;
> +    remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
> +                                              header);
> +  }
> +
>    return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
>  }

Patch failure is because a revert was landed already. See 
https://codereview.webrtc.org/2668973003/

You should probably close this issue.

Powered by Google App Engine
This is Rietveld 408576698