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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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