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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Issue 1338203003: Wire up send-side bandwidth estimation. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments, 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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75 configuration.audio, 75 configuration.audio,
76 configuration.clock, 76 configuration.clock,
77 configuration.receive_statistics, 77 configuration.receive_statistics,
78 configuration.rtcp_packet_type_counter_observer), 78 configuration.rtcp_packet_type_counter_observer),
79 rtcp_receiver_(configuration.id, 79 rtcp_receiver_(configuration.id,
80 configuration.clock, 80 configuration.clock,
81 configuration.receiver_only, 81 configuration.receiver_only,
82 configuration.rtcp_packet_type_counter_observer, 82 configuration.rtcp_packet_type_counter_observer,
83 configuration.bandwidth_callback, 83 configuration.bandwidth_callback,
84 configuration.intra_frame_callback, 84 configuration.intra_frame_callback,
85 configuration.transport_feedback_callback,
85 this), 86 this),
86 clock_(configuration.clock), 87 clock_(configuration.clock),
87 id_(configuration.id), 88 id_(configuration.id),
88 audio_(configuration.audio), 89 audio_(configuration.audio),
89 collision_detected_(false), 90 collision_detected_(false),
90 last_process_time_(configuration.clock->TimeInMilliseconds()), 91 last_process_time_(configuration.clock->TimeInMilliseconds()),
91 last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()), 92 last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
92 last_rtt_process_time_(configuration.clock->TimeInMilliseconds()), 93 last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
93 packet_overhead_(28), // IPV4 UDP. 94 packet_overhead_(28), // IPV4 UDP.
94 padding_index_(static_cast<size_t>(-1)), // Start padding at first child. 95 padding_index_(static_cast<size_t>(-1)), // Start padding at first child.
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993 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 994 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
994 StreamDataCountersCallback* callback) { 995 StreamDataCountersCallback* callback) {
995 rtp_sender_.RegisterRtpStatisticsCallback(callback); 996 rtp_sender_.RegisterRtpStatisticsCallback(callback);
996 } 997 }
997 998
998 StreamDataCountersCallback* 999 StreamDataCountersCallback*
999 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 1000 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
1000 return rtp_sender_.GetRtpStatisticsCallback(); 1001 return rtp_sender_.GetRtpStatisticsCallback();
1001 } 1002 }
1002 } // namespace webrtc 1003 } // namespace webrtc
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