webrtc/modules/rtp_rtcp/include/rtp_rtcp.h - Issue 2061423003: Refactor NACK bitrate allocation

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Issue 2061423003: Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Moved rate limiter and addressed comments Created 4 years, 5 months ago

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« webrtc/modules/pacing/paced_sender.cc ('K') | « webrtc/modules/rtp_rtcp/include/receive_statistics.h ('k') | webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h » ('j') | webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h » ('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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72 RtcpRttStats* rtt_stats;	72 RtcpRttStats* rtt_stats;

73 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;	73 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;

74 RemoteBitrateEstimator* remote_bitrate_estimator;	74 RemoteBitrateEstimator* remote_bitrate_estimator;

75 RtpPacketSender* paced_sender;	75 RtpPacketSender* paced_sender;

76 TransportSequenceNumberAllocator* transport_sequence_number_allocator;	76 TransportSequenceNumberAllocator* transport_sequence_number_allocator;

77 BitrateStatisticsObserver* send_bitrate_observer;	77 BitrateStatisticsObserver* send_bitrate_observer;

78 FrameCountObserver* send_frame_count_observer;	78 FrameCountObserver* send_frame_count_observer;

79 SendSideDelayObserver* send_side_delay_observer;	79 SendSideDelayObserver* send_side_delay_observer;

80 RtcEventLog* event_log;	80 RtcEventLog* event_log;

81 SendPacketObserver* send_packet_observer;	81 SendPacketObserver* send_packet_observer;

	82 NackRateLimiter* nack_rate_limiter;

82 RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);	83 RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);

83 };	84 };

84	85

85 /*	86 /*

86 * Create a RTP/RTCP module object using the system clock.	87 * Create a RTP/RTCP module object using the system clock.

87 *	88 *

88 * configuration - Configuration of the RTP/RTCP module.	89 * configuration - Configuration of the RTP/RTCP module.

89 */	90 */

90 static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);	91 static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);

91	92

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647	648

648 /*	649 /*

649 * send a request for a keyframe	650 * send a request for a keyframe

650 *	651 *

651 * return -1 on failure else 0	652 * return -1 on failure else 0

652 */	653 */

653 virtual int32_t RequestKeyFrame() = 0;	654 virtual int32_t RequestKeyFrame() = 0;

654 };	655 };

655 } // namespace webrtc	656 } // namespace webrtc

656 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_	657 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_

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