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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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77 #include "webrtc/api/datachannelinterface.h" | 77 #include "webrtc/api/datachannelinterface.h" |
78 #include "webrtc/api/dtmfsenderinterface.h" | 78 #include "webrtc/api/dtmfsenderinterface.h" |
79 #include "webrtc/api/jsep.h" | 79 #include "webrtc/api/jsep.h" |
80 #include "webrtc/api/mediastreaminterface.h" | 80 #include "webrtc/api/mediastreaminterface.h" |
81 #include "webrtc/api/rtcerror.h" | 81 #include "webrtc/api/rtcerror.h" |
82 #include "webrtc/api/rtpreceiverinterface.h" | 82 #include "webrtc/api/rtpreceiverinterface.h" |
83 #include "webrtc/api/rtpsenderinterface.h" | 83 #include "webrtc/api/rtpsenderinterface.h" |
84 #include "webrtc/api/stats/rtcstatscollectorcallback.h" | 84 #include "webrtc/api/stats/rtcstatscollectorcallback.h" |
85 #include "webrtc/api/statstypes.h" | 85 #include "webrtc/api/statstypes.h" |
86 #include "webrtc/api/umametrics.h" | 86 #include "webrtc/api/umametrics.h" |
87 #include "webrtc/base/bitrateallocationstrategy.h" | |
87 #include "webrtc/base/fileutils.h" | 88 #include "webrtc/base/fileutils.h" |
88 #include "webrtc/base/network.h" | 89 #include "webrtc/base/network.h" |
89 #include "webrtc/base/rtccertificate.h" | 90 #include "webrtc/base/rtccertificate.h" |
90 #include "webrtc/base/rtccertificategenerator.h" | 91 #include "webrtc/base/rtccertificategenerator.h" |
91 #include "webrtc/base/socketaddress.h" | 92 #include "webrtc/base/socketaddress.h" |
92 #include "webrtc/base/sslstreamadapter.h" | 93 #include "webrtc/base/sslstreamadapter.h" |
93 #include "webrtc/call/callfactoryinterface.h" | 94 #include "webrtc/call/callfactoryinterface.h" |
94 #include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h" | 95 #include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h" |
95 #include "webrtc/media/base/mediachannel.h" | 96 #include "webrtc/media/base/mediachannel.h" |
96 #include "webrtc/media/base/videocapturer.h" | 97 #include "webrtc/media/base/videocapturer.h" |
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752 // SetBitrate limits the bandwidth allocated for all RTP streams sent by | 753 // SetBitrate limits the bandwidth allocated for all RTP streams sent by |
753 // this PeerConnection. Other limitations might affect these limits and | 754 // this PeerConnection. Other limitations might affect these limits and |
754 // are respected (for example "b=AS" in SDP). | 755 // are respected (for example "b=AS" in SDP). |
755 // | 756 // |
756 // Setting |current_bitrate_bps| will reset the current bitrate estimate | 757 // Setting |current_bitrate_bps| will reset the current bitrate estimate |
757 // to the provided value. | 758 // to the provided value. |
758 virtual RTCError SetBitrate(const BitrateParameters& bitrate) { | 759 virtual RTCError SetBitrate(const BitrateParameters& bitrate) { |
759 return RTCError::OK(); | 760 return RTCError::OK(); |
760 } | 761 } |
761 | 762 |
763 // SetBitrateAllocationStrategy sets current strategy. If not set default | |
764 // WEBRTC allocator will be used. May be changed during an active session. | |
765 virtual RTCError SetBitrateAllocationStrategy( | |
stefan-webrtc
2017/08/14 07:46:03
I would prefer if we could pass the allocation str
alexnarest
2017/08/15 06:02:36
Helper would let users to change strategies during
| |
766 rtc::BitrateAllocationStrategy* bitrate_allocation_strategy) { | |
767 return RTCError(RTCErrorType::UNSUPPORTED_OPERATION); | |
768 } | |
769 | |
762 // Returns the current SignalingState. | 770 // Returns the current SignalingState. |
763 virtual SignalingState signaling_state() = 0; | 771 virtual SignalingState signaling_state() = 0; |
764 virtual IceConnectionState ice_connection_state() = 0; | 772 virtual IceConnectionState ice_connection_state() = 0; |
765 virtual IceGatheringState ice_gathering_state() = 0; | 773 virtual IceGatheringState ice_gathering_state() = 0; |
766 | 774 |
767 // Starts RtcEventLog using existing file. Takes ownership of |file| and | 775 // Starts RtcEventLog using existing file. Takes ownership of |file| and |
768 // passes it on to Call, which will take the ownership. If the | 776 // passes it on to Call, which will take the ownership. If the |
769 // operation fails the file will be closed. The logging will stop | 777 // operation fails the file will be closed. The logging will stop |
770 // automatically after 10 minutes have passed, or when the StopRtcEventLog | 778 // automatically after 10 minutes have passed, or when the StopRtcEventLog |
771 // function is called. | 779 // function is called. |
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1180 cricket::WebRtcVideoEncoderFactory* video_encoder_factory, | 1188 cricket::WebRtcVideoEncoderFactory* video_encoder_factory, |
1181 cricket::WebRtcVideoDecoderFactory* video_decoder_factory, | 1189 cricket::WebRtcVideoDecoderFactory* video_decoder_factory, |
1182 rtc::scoped_refptr<AudioMixer> audio_mixer, | 1190 rtc::scoped_refptr<AudioMixer> audio_mixer, |
1183 std::unique_ptr<cricket::MediaEngineInterface> media_engine, | 1191 std::unique_ptr<cricket::MediaEngineInterface> media_engine, |
1184 std::unique_ptr<CallFactoryInterface> call_factory, | 1192 std::unique_ptr<CallFactoryInterface> call_factory, |
1185 std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory); | 1193 std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory); |
1186 | 1194 |
1187 } // namespace webrtc | 1195 } // namespace webrtc |
1188 | 1196 |
1189 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 1197 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
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