| Index: webrtc/api/peerconnectioninterface.h
|
| diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
|
| index 64f5b2da6ec320862e5b478b1d3a2e92b3293c3e..8c5a64bf1ebf69d197ac61d55c1731e95bdf7699 100644
|
| --- a/webrtc/api/peerconnectioninterface.h
|
| +++ b/webrtc/api/peerconnectioninterface.h
|
| @@ -729,21 +729,6 @@
|
| // destroyed, RegisterUMAOberver(nullptr) should be called.
|
| virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
|
|
|
| - // 0 <= min <= current <= max should hold for set parameters.
|
| - 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).
|
| - //
|
| - // Setting |current_bitrate_bps| will reset the current 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;
|
|
|