Chromium Code Reviews| Index: webrtc/api/peerconnectioninterface.h |
| diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h |
| index 8c5a64bf1ebf69d197ac61d55c1731e95bdf7699..8bfd4d28284db2c0339362bb868e2929664103cf 100644 |
| --- a/webrtc/api/peerconnectioninterface.h |
| +++ b/webrtc/api/peerconnectioninterface.h |
| @@ -729,6 +729,21 @@ class PeerConnectionInterface : public rtc::RefCountInterface { |
| // destroyed, RegisterUMAOberver(nullptr) should be called. |
| virtual void RegisterUMAObserver(UMAObserver* observer) = 0; |
| + // 0 <= min <= current <= max should hold for set parameters. |
|
kwiberg-webrtc
2017/05/24 08:27:55
Are all 8 combinations of set/unset supported?
Zach Stein
2017/05/24 21:10:02
Yes.
|
| + struct BitrateParameters { |
| + rtc::Optional<int> min_bitrate_bps; |
| + rtc::Optional<int> current_bitrate_bps; |
| + rtc::Optional<int> max_bitrate_bps; |
| + }; |
| + |
| + // SetBitrate limits the bandwidth allocated for all RTP streams sent by |
| + // this PeerConnection. Other limitations might affect these limits and |
| + // are respected (for example "b=AS" in SDP). |
| + // |
| + // Changing |current_bitrate_bps| to a new value will reset the current |
|
Taylor Brandstetter
2017/05/25 15:33:52
This comment may need to be updated. It's not just
Zach Stein
2017/05/25 20:26:31
Done.
|
| + // bitrate estimate to the provided value. |
| + virtual RTCError SetBitrate(const BitrateParameters& bitrate) = 0; |
| + |
| // Returns the current SignalingState. |
| virtual SignalingState signaling_state() = 0; |
| virtual IceConnectionState ice_connection_state() = 0; |