webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h - Issue 1329083005: Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor

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Issue 1329083005: Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Created 5 years, 3 months ago

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« webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc ('K') | « webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc ('k') | webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h » ('j') | webrtc/video_engine/vie_channel_group.cc » ('J')
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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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55 * bursts to minimize packet loss.	55 * bursts to minimize packet loss.

56 */	56 */

57 int32_t id;	57 int32_t id;

58 bool audio;	58 bool audio;

59 bool receiver_only;	59 bool receiver_only;

60 Clock* clock;	60 Clock* clock;

61 ReceiveStatistics* receive_statistics;	61 ReceiveStatistics* receive_statistics;

62 Transport* outgoing_transport;	62 Transport* outgoing_transport;

63 RtcpIntraFrameObserver* intra_frame_callback;	63 RtcpIntraFrameObserver* intra_frame_callback;

64 RtcpBandwidthObserver* bandwidth_callback;	64 RtcpBandwidthObserver* bandwidth_callback;

65 SendTimeObserver* send_time_callback;	65 TransportFeedbackObserver* transport_feedback_callback;

66 RtcpRttStats* rtt_stats;	66 RtcpRttStats* rtt_stats;

67 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;	67 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;

68 RtpAudioFeedback* audio_messages;	68 RtpAudioFeedback* audio_messages;

69 RemoteBitrateEstimator* remote_bitrate_estimator;	69 RemoteBitrateEstimator* remote_bitrate_estimator;

70 PacedSender* paced_sender;	70 PacedSender* paced_sender;

71 PacketRouter* packet_router;	71 PacketRouter* packet_router;

72 BitrateStatisticsObserver* send_bitrate_observer;	72 BitrateStatisticsObserver* send_bitrate_observer;

73 FrameCountObserver* send_frame_count_observer;	73 FrameCountObserver* send_frame_count_observer;

74 SendSideDelayObserver* send_side_delay_observer;	74 SendSideDelayObserver* send_side_delay_observer;

75 };	75 };

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630	630

631 /*	631 /*

632 * send a request for a keyframe	632 * send a request for a keyframe

633 *	633 *

634 * return -1 on failure else 0	634 * return -1 on failure else 0

635 */	635 */

636 virtual int32_t RequestKeyFrame() = 0;	636 virtual int32_t RequestKeyFrame() = 0;

637 };	637 };

638 } // namespace webrtc	638 } // namespace webrtc

639 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_	639 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_

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