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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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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/call/callfactoryinterface.h" | 87 #include "webrtc/call/callfactoryinterface.h" |
88 #include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h" | 88 #include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h" |
89 #include "webrtc/media/base/mediachannel.h" | 89 #include "webrtc/media/base/mediachannel.h" |
90 #include "webrtc/media/base/videocapturer.h" | 90 #include "webrtc/media/base/videocapturer.h" |
91 #include "webrtc/p2p/base/portallocator.h" | 91 #include "webrtc/p2p/base/portallocator.h" |
| 92 #include "webrtc/rtc_base/bitrateallocationstrategy.h" |
92 #include "webrtc/rtc_base/fileutils.h" | 93 #include "webrtc/rtc_base/fileutils.h" |
93 #include "webrtc/rtc_base/network.h" | 94 #include "webrtc/rtc_base/network.h" |
94 #include "webrtc/rtc_base/rtccertificate.h" | 95 #include "webrtc/rtc_base/rtccertificate.h" |
95 #include "webrtc/rtc_base/rtccertificategenerator.h" | 96 #include "webrtc/rtc_base/rtccertificategenerator.h" |
96 #include "webrtc/rtc_base/socketaddress.h" | 97 #include "webrtc/rtc_base/socketaddress.h" |
97 #include "webrtc/rtc_base/sslstreamadapter.h" | 98 #include "webrtc/rtc_base/sslstreamadapter.h" |
98 | 99 |
99 namespace rtc { | 100 namespace rtc { |
100 class SSLIdentity; | 101 class SSLIdentity; |
101 class Thread; | 102 class Thread; |
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764 }; | 765 }; |
765 | 766 |
766 // SetBitrate limits the bandwidth allocated for all RTP streams sent by | 767 // SetBitrate limits the bandwidth allocated for all RTP streams sent by |
767 // this PeerConnection. Other limitations might affect these limits and | 768 // this PeerConnection. Other limitations might affect these limits and |
768 // are respected (for example "b=AS" in SDP). | 769 // are respected (for example "b=AS" in SDP). |
769 // | 770 // |
770 // Setting |current_bitrate_bps| will reset the current bitrate estimate | 771 // Setting |current_bitrate_bps| will reset the current bitrate estimate |
771 // to the provided value. | 772 // to the provided value. |
772 virtual RTCError SetBitrate(const BitrateParameters& bitrate) = 0; | 773 virtual RTCError SetBitrate(const BitrateParameters& bitrate) = 0; |
773 | 774 |
| 775 // SetBitrateAllocationStrategy sets current strategy. If not set default |
| 776 // WebRTC allocator will be used. May be changed during an active session. |
| 777 // Should be set to null before the strategy is destroyed. The strategy is |
| 778 // owned by application and it is responsible for keeping it alive as long as |
| 779 // any of its peerconnections are using it. |
| 780 virtual RTCError SetBitrateAllocationStrategy( |
| 781 rtc::BitrateAllocationStrategy* bitrate_allocation_strategy) { |
| 782 return RTCError(RTCErrorType::UNSUPPORTED_OPERATION); |
| 783 } |
| 784 |
774 // Returns the current SignalingState. | 785 // Returns the current SignalingState. |
775 virtual SignalingState signaling_state() = 0; | 786 virtual SignalingState signaling_state() = 0; |
776 virtual IceConnectionState ice_connection_state() = 0; | 787 virtual IceConnectionState ice_connection_state() = 0; |
777 virtual IceGatheringState ice_gathering_state() = 0; | 788 virtual IceGatheringState ice_gathering_state() = 0; |
778 | 789 |
779 // Starts RtcEventLog using existing file. Takes ownership of |file| and | 790 // Starts RtcEventLog using existing file. Takes ownership of |file| and |
780 // passes it on to Call, which will take the ownership. If the | 791 // passes it on to Call, which will take the ownership. If the |
781 // operation fails the file will be closed. The logging will stop | 792 // operation fails the file will be closed. The logging will stop |
782 // automatically after 10 minutes have passed, or when the StopRtcEventLog | 793 // automatically after 10 minutes have passed, or when the StopRtcEventLog |
783 // function is called. | 794 // function is called. |
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1212 cricket::WebRtcVideoEncoderFactory* video_encoder_factory, | 1223 cricket::WebRtcVideoEncoderFactory* video_encoder_factory, |
1213 cricket::WebRtcVideoDecoderFactory* video_decoder_factory, | 1224 cricket::WebRtcVideoDecoderFactory* video_decoder_factory, |
1214 rtc::scoped_refptr<AudioMixer> audio_mixer, | 1225 rtc::scoped_refptr<AudioMixer> audio_mixer, |
1215 std::unique_ptr<cricket::MediaEngineInterface> media_engine, | 1226 std::unique_ptr<cricket::MediaEngineInterface> media_engine, |
1216 std::unique_ptr<CallFactoryInterface> call_factory, | 1227 std::unique_ptr<CallFactoryInterface> call_factory, |
1217 std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory); | 1228 std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory); |
1218 | 1229 |
1219 } // namespace webrtc | 1230 } // namespace webrtc |
1220 | 1231 |
1221 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 1232 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
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