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Issue 2978503002: Move RTP keep-alive config from VideoSendStream::Config to Call::Config (Closed)

Created:
3 years, 5 months ago by sprang_webrtc
Modified:
3 years, 5 months ago
Reviewers:
pbos-webrtc
CC:
webrtc-reviews_webrtc.org, interface-changes_webrtc.org, video-team_agora.io, yujie_mao (webrtc), zhengzhonghou_agora.io, stefan-webrtc, tterriberry_mozilla.com, qiang.lu, niklas.enbom, peah-webrtc, the sun, mflodman
Target Ref:
refs/heads/master
Project:
webrtc
Visibility:
Public.

Description

Move RTP keep-alive config from VideoSendStream::Config to Call::Config This makes more sense since logically it's a transport level feature, not a media stream feature. Even if the implementation details forces it to be an rtp stream detail, for the moment. BUG=webrtc:7907 Review-Url: https://codereview.webrtc.org/2978503002 Cr-Commit-Position: refs/heads/master@{#18963} Committed: https://chromium.googlesource.com/external/webrtc/+/e5c4a810e21e74acec09c987f20fb0e6788062fd

Patch Set 1 #

Total comments: 2

Patch Set 2 : Comment #

Patch Set 3 : Typo in comment #

Unified diffs Side-by-side diffs Delta from patch set Stats (+32 lines, -21 lines) Patch
M webrtc/call/call.h View 1 2 1 chunk +6 lines, -0 lines 0 comments Download
M webrtc/call/call.cc View 1 chunk +2 lines, -1 line 0 comments Download
M webrtc/video/video_send_stream.h View 1 chunk +2 lines, -1 line 0 comments Download
M webrtc/video/video_send_stream.cc View 9 chunks +14 lines, -8 lines 0 comments Download
M webrtc/video/video_send_stream_tests.cc View 2 chunks +8 lines, -9 lines 0 comments Download
M webrtc/video_send_stream.h View 1 chunk +0 lines, -2 lines 0 comments Download

Messages

Total messages: 14 (9 generated)
sprang_webrtc
3 years, 5 months ago (2017-07-10 15:49:06 UTC) #2
pbos-webrtc
lgtm https://codereview.webrtc.org/2978503002/diff/1/webrtc/call/call.h File webrtc/call/call.h (right): https://codereview.webrtc.org/2978503002/diff/1/webrtc/call/call.h#newcode117 webrtc/call/call.h:117: // network time-out events. Is there a RFC ...
3 years, 5 months ago (2017-07-10 22:40:24 UTC) #3
sprang_webrtc
https://codereview.webrtc.org/2978503002/diff/1/webrtc/call/call.h File webrtc/call/call.h (right): https://codereview.webrtc.org/2978503002/diff/1/webrtc/call/call.h#newcode117 webrtc/call/call.h:117: // network time-out events. On 2017/07/10 22:40:23, pbos-webrtc wrote: ...
3 years, 5 months ago (2017-07-11 08:39:58 UTC) #4
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/2978503002/40001
3 years, 5 months ago (2017-07-11 10:41:37 UTC) #11
commit-bot: I haz the power
3 years, 5 months ago (2017-07-11 10:44:23 UTC) #14
Message was sent while issue was closed.
Committed patchset #3 (id:40001) as
https://chromium.googlesource.com/external/webrtc/+/e5c4a810e21e74acec09c987f...

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