| commit | bcf91808a2205e72e9a669ff80f6fe9c15f6d349 | [log] [tgz] |
|---|---|---|
| author | Alex Narest <alexnarest@webrtc.org> | Mon Jun 25 14:08:36 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Jun 27 10:33:40 2018 |
| tree | 7a697368a51e46c5ef09b860eb3da906507c33d0 | |
| parent | 81f51975125c6c12b6dc9fd470132c78ef4812c9 [diff] [blame] |
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. 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}
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h index 0027696..d9a4e18 100644 --- a/call/rtp_transport_controller_send.h +++ b/call/rtp_transport_controller_send.h
@@ -85,6 +85,8 @@ 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_;