Commit graph

6 commits

Author SHA1 Message Date
Christoffer Rodbro
110c64bcd6 Delete unused key WebRTC-Audio-SendSideBwe-For-Video.
Bug: webrtc:10286
Change-Id: If9ddbe71d9ba1afe51be5f9f46fcd4a72b34bc7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123787
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26990}
2019-03-06 13:15:53 +00:00
Per Kjellander
914351de5c Reland "Always offer transport sequence number header extension for audio""
(reverted in https://webrtc-review.googlesource.com/c/src/+/123182/1)

Original cl description:
Always offer transport sequence number header extension for audio

If the extension is negotiated, it will only be used if
the field trial WebRTC-Audio-SendSideBwe is enabled.
This allows simpler experimentation if it should be used or not.

Patchset 3 contain the only change:
  Add the field trial WebRTC-Audio-SendSideBwe to  call/rampup_tests.cc

TBR: srte@webrtc.org,ossu@webrtc.org
Bug: webrtc:10309 webrtc:10286
Change-Id: I2c1224e8a9fab52c1030369c1364686322e88a0f
Reviewed-on: https://webrtc-review.googlesource.com/c/123183
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26706}
2019-02-15 10:57:38 +00:00
Ying Wang
397c06fe9d Revert "Always offer transport sequence number header extension for audio"
This reverts commit fd965c008c.

Reason for revert: Cause test failure.

Original change's description:
> Always offer transport sequence number header extension for audio
> 
> If the extension is negotiated, it will only be used if
> the field trial WebRTC-Audio-SendSideBwe is enabled.
> This allows simpler experimentation if it should be used or not.
> 
> Bug: webrtc:10309 webrtc:10286
> Change-Id: I797e6f14c06d46189e40f6d09805c2e09afc015b
> Reviewed-on: https://webrtc-review.googlesource.com/c/122542
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26689}

TBR=ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org

Change-Id: I1b7d3fa5c282a5bf049ca54695ad16c8278a2698
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10309 webrtc:10286
Reviewed-on: https://webrtc-review.googlesource.com/c/123182
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26703}
2019-02-15 08:53:25 +00:00
Per Kjellander
fd965c008c Always offer transport sequence number header extension for audio
If the extension is negotiated, it will only be used if
the field trial WebRTC-Audio-SendSideBwe is enabled.
This allows simpler experimentation if it should be used or not.

Bug: webrtc:10309 webrtc:10286
Change-Id: I797e6f14c06d46189e40f6d09805c2e09afc015b
Reviewed-on: https://webrtc-review.googlesource.com/c/122542
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26689}
2019-02-14 15:28:07 +00:00
Sebastian Jansson
464a5576ea Adds audio priority bitrate field trial parameter.
This replaces the functionality provided by
AudioPriorityBitrateAllocationStrategy, removing the need provide that
component via injection in all clients using audio bitrate priority.

Bug: webrtc:10286
Change-Id: I3bafab56d24459d9d27dc07abffdc8538440a346
Reviewed-on: https://webrtc-review.googlesource.com/c/121402
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26651}
2019-02-12 16:03:22 +00:00
Sebastian Jansson
470a5eae93 Introduces common AudioAllocationSettings class.
This class collects the field trial based configuration of audio
allocation and bandwidth in one place. This makes it easier
overview and prepares for future cleanup of the trials.

Bug: webrtc:9718
Change-Id: I34a441c0165b423f1e2ee63894337484684146ac
Reviewed-on: https://webrtc-review.googlesource.com/c/118282
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26370}
2019-01-23 12:13:29 +00:00