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Side by Side Diff: webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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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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