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Side by Side Diff: webrtc/api/peerconnectioninterface.h

Issue 2924243003: Remove default implementation of PeerConnectionInterface::SetBitrate. (Closed)
Patch Set: Created 3 years, 6 months ago
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1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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735 rtc::Optional<int> current_bitrate_bps; 735 rtc::Optional<int> current_bitrate_bps;
736 rtc::Optional<int> max_bitrate_bps; 736 rtc::Optional<int> max_bitrate_bps;
737 }; 737 };
738 738
739 // SetBitrate limits the bandwidth allocated for all RTP streams sent by 739 // SetBitrate limits the bandwidth allocated for all RTP streams sent by
740 // this PeerConnection. Other limitations might affect these limits and 740 // this PeerConnection. Other limitations might affect these limits and
741 // are respected (for example "b=AS" in SDP). 741 // are respected (for example "b=AS" in SDP).
742 // 742 //
743 // Setting |current_bitrate_bps| will reset the current bitrate estimate 743 // Setting |current_bitrate_bps| will reset the current bitrate estimate
744 // to the provided value. 744 // to the provided value.
745 virtual RTCError SetBitrate(const BitrateParameters& bitrate) { 745 virtual RTCError SetBitrate(const BitrateParameters& bitrate) = 0;
746 return RTCError::OK();
747 }
748 746
749 // Returns the current SignalingState. 747 // Returns the current SignalingState.
750 virtual SignalingState signaling_state() = 0; 748 virtual SignalingState signaling_state() = 0;
751 virtual IceConnectionState ice_connection_state() = 0; 749 virtual IceConnectionState ice_connection_state() = 0;
752 virtual IceGatheringState ice_gathering_state() = 0; 750 virtual IceGatheringState ice_gathering_state() = 0;
753 751
754 // Starts RtcEventLog using existing file. Takes ownership of |file| and 752 // Starts RtcEventLog using existing file. Takes ownership of |file| and
755 // passes it on to Call, which will take the ownership. If the 753 // passes it on to Call, which will take the ownership. If the
756 // operation fails the file will be closed. The logging will stop 754 // operation fails the file will be closed. The logging will stop
757 // automatically after 10 minutes have passed, or when the StopRtcEventLog 755 // automatically after 10 minutes have passed, or when the StopRtcEventLog
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1122 cricket::WebRtcVideoEncoderFactory* encoder_factory, 1120 cricket::WebRtcVideoEncoderFactory* encoder_factory,
1123 cricket::WebRtcVideoDecoderFactory* decoder_factory) { 1121 cricket::WebRtcVideoDecoderFactory* decoder_factory) {
1124 return CreatePeerConnectionFactory( 1122 return CreatePeerConnectionFactory(
1125 worker_and_network_thread, worker_and_network_thread, signaling_thread, 1123 worker_and_network_thread, worker_and_network_thread, signaling_thread,
1126 default_adm, encoder_factory, decoder_factory); 1124 default_adm, encoder_factory, decoder_factory);
1127 } 1125 }
1128 1126
1129 } // namespace webrtc 1127 } // namespace webrtc
1130 1128
1131 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ 1129 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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