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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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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50 * intra_frame_callback - Called when the receiver request a intra frame. | 50 * intra_frame_callback - Called when the receiver request a intra frame. |
51 * bandwidth_callback - Called when we receive a changed estimate from | 51 * bandwidth_callback - Called when we receive a changed estimate from |
52 * the receiver of out stream. | 52 * the receiver of out stream. |
53 * audio_messages - Telephone events. May not be NULL; default | 53 * audio_messages - Telephone events. May not be NULL; default |
54 * callback will do nothing. | 54 * callback will do nothing. |
55 * remote_bitrate_estimator - Estimates the bandwidth available for a set of | 55 * remote_bitrate_estimator - Estimates the bandwidth available for a set of |
56 * streams from the same client. | 56 * streams from the same client. |
57 * paced_sender - Spread any bursts of packets into smaller | 57 * paced_sender - Spread any bursts of packets into smaller |
58 * bursts to minimize packet loss. | 58 * bursts to minimize packet loss. |
59 */ | 59 */ |
60 int32_t id; | |
61 bool audio; | 60 bool audio; |
62 bool receiver_only; | 61 bool receiver_only; |
63 Clock* clock; | 62 Clock* clock; |
64 ReceiveStatistics* receive_statistics; | 63 ReceiveStatistics* receive_statistics; |
65 Transport* outgoing_transport; | 64 Transport* outgoing_transport; |
66 RtcpIntraFrameObserver* intra_frame_callback; | 65 RtcpIntraFrameObserver* intra_frame_callback; |
67 RtcpBandwidthObserver* bandwidth_callback; | 66 RtcpBandwidthObserver* bandwidth_callback; |
68 TransportFeedbackObserver* transport_feedback_callback; | 67 TransportFeedbackObserver* transport_feedback_callback; |
69 RtcpRttStats* rtt_stats; | 68 RtcpRttStats* rtt_stats; |
70 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer; | 69 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer; |
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635 | 634 |
636 /* | 635 /* |
637 * send a request for a keyframe | 636 * send a request for a keyframe |
638 * | 637 * |
639 * return -1 on failure else 0 | 638 * return -1 on failure else 0 |
640 */ | 639 */ |
641 virtual int32_t RequestKeyFrame() = 0; | 640 virtual int32_t RequestKeyFrame() = 0; |
642 }; | 641 }; |
643 } // namespace webrtc | 642 } // namespace webrtc |
644 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ | 643 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ |
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