[Remoting Host] Stop using SDP to set audio bandwidth |
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Issue descriptionPer discussion with WebRTC members, the CRD host is modifying the local SDP to adjust the audio bandwidth being used. We were advised that this approach will stop working at some point in the future and we should set our Audio bandwidth via a supported method (suggestion was to use RTPSender.setParameters() via the RTCRTPEncodingParameters.maxBitrate parameter).
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Nov 8
We should stop modifying SDP altogether (including the x-google-[min/max]-bitrate settings). This will help us migrate to Unified-Plan.
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Nov 8
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Jan 16
The following revision refers to this bug: https://chromium.googlesource.com/chromium/src.git/+/3fc40106b0c32d06ac3d5ebc59a78565263c6ec6 commit 3fc40106b0c32d06ac3d5ebc59a78565263c6ec6 Author: Lambros Lambrou <lambroslambrou@chromium.org> Date: Wed Jan 16 01:56:56 2019 Set bitrate caps using newer APIs This removes the x-google-[min/max]-bitrate SDP parameters used to set the bandwidth caps for the connection. Instead, the global caps are set using PeerConnection::SetBitrate(), and the VideoSender caps are set to the same values using RtpSender::SetParameters(), which was already being used to set the max-bitrate cap for relay connections. It appears that both global and per-sender caps are needed to properly configure the b/w estimator and ALR probers, otherwise the bitrate is severely constrained by WebRTC defaults (~600kbps). This CL unblocks the UnifiedPlan migration, and ensures we are using consistent APIs to set the bitrate caps for both direct and relayed connections. Bug: 903012 , 833538 Change-Id: Iefaddf877071f0fc3a010cf02a2b1fd2a92ad00c Reviewed-on: https://chromium-review.googlesource.com/c/1409710 Reviewed-by: Joe Downing <joedow@chromium.org> Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org> Cr-Commit-Position: refs/heads/master@{#623014} [modify] https://crrev.com/3fc40106b0c32d06ac3d5ebc59a78565263c6ec6/remoting/protocol/webrtc_transport.cc
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Jan 17
(6 days ago)
The host no longer uses SDP to set video bitrates. That is sufficient to unblock UnifiedPlan migration. However, the host and client both add audio parameters: Host adds to SDP offer and incoming answer: "stereo=1; maxaveragebitrate=163840" Client adds to SDP outgoing answer: "stereo=1; x-google-min-bitrate=160" Together, they both cause the Opus encoder to output stereo at 160kbps. I verified this still works with UnifiedPlan. It would still be nicer to use RtpSender::SetParameters() to hardcode these values (like we do for the Chrome app), instead of SDP munging and negotiation. We've been told that SDP munging might stop working at some point, so we want to do this work.
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Jan 17
(6 days ago)
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Comment 1 by jamiewa...@chromium.org
, May 15 2018Owner: lambroslambrou@chromium.org
Status: Assigned (was: Untriaged)