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Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.

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This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead.

Bug: webrtc:8243
Change-Id: If961780921d53bdce95b68c26641df6875509c1f
Reviewed-on: https://webrtc-review.googlesource.com/84501
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23755}
This commit is contained in:
Alex Narest
2018-06-25 16:08:36 +02:00
committed by Commit Bot
parent 81f5197512
commit bcf91808a2
13 changed files with 46 additions and 3 deletions
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2
call/rtp_transport_controller_send.h
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@ -85,6 +85,8 @@ class RtpTransportControllerSend final
void SetSdpBitrateParameters(const BitrateConstraints& constraints) override;
void SetClientBitratePreferences(const BitrateSettings& preferences) override;
void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) override;
private:
const Clock* const clock_;
PacketRouter packet_router_;