webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc - Issue 1623543002: Refactor RtpSender and SSRCDatabase a bit.

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Issue 1623543002: Refactor RtpSender and SSRCDatabase a bit. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Add TODOs for Random() and delete dead code from Timing (move to separate cl?) Created 4 years, 11 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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63 : rtp_sender_(configuration.audio,	63 : rtp_sender_(configuration.audio,

64 configuration.clock,	64 configuration.clock,

65 configuration.outgoing_transport,	65 configuration.outgoing_transport,

66 configuration.audio_messages,	66 configuration.audio_messages,

67 configuration.paced_sender,	67 configuration.paced_sender,

68 configuration.transport_sequence_number_allocator,	68 configuration.transport_sequence_number_allocator,

69 configuration.transport_feedback_callback,	69 configuration.transport_feedback_callback,

70 configuration.send_bitrate_observer,	70 configuration.send_bitrate_observer,

71 configuration.send_frame_count_observer,	71 configuration.send_frame_count_observer,

72 configuration.send_side_delay_observer,	72 configuration.send_side_delay_observer,

73 configuration.event_log),	73 configuration.event_log,

	74 &ssrc_database_),

74 rtcp_sender_(configuration.audio,	75 rtcp_sender_(configuration.audio,

75 configuration.clock,	76 configuration.clock,

76 configuration.receive_statistics,	77 configuration.receive_statistics,

77 configuration.rtcp_packet_type_counter_observer,	78 configuration.rtcp_packet_type_counter_observer,

78 configuration.event_log,	79 configuration.event_log,

79 configuration.outgoing_transport),	80 configuration.outgoing_transport),

80 rtcp_receiver_(configuration.clock,	81 rtcp_receiver_(configuration.clock,

81 configuration.receiver_only,	82 configuration.receiver_only,

82 configuration.rtcp_packet_type_counter_observer,	83 configuration.rtcp_packet_type_counter_observer,

83 configuration.bandwidth_callback,	84 configuration.bandwidth_callback,

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992 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(	993 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(

993 StreamDataCountersCallback* callback) {	994 StreamDataCountersCallback* callback) {

994 rtp_sender_.RegisterRtpStatisticsCallback(callback);	995 rtp_sender_.RegisterRtpStatisticsCallback(callback);

995 }	996 }

996	997

997 StreamDataCountersCallback*	998 StreamDataCountersCallback*

998 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {	999 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {

999 return rtp_sender_.GetRtpStatisticsCallback();	1000 return rtp_sender_.GetRtpStatisticsCallback();

1000 }	1001 }

1001 } // namespace webrtc	1002 } // namespace webrtc

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