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

Issue 2032463003: Revert of Propagate probing cluster id to SendTimeHistory. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 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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421 rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); 421 rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
422 } 422 }
423 return rtp_sender_.SendOutgoingData( 423 return rtp_sender_.SendOutgoingData(
424 frame_type, payload_type, time_stamp, capture_time_ms, payload_data, 424 frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
425 payload_size, fragmentation, rtp_video_hdr); 425 payload_size, fragmentation, rtp_video_hdr);
426 } 426 }
427 427
428 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, 428 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
429 uint16_t sequence_number, 429 uint16_t sequence_number,
430 int64_t capture_time_ms, 430 int64_t capture_time_ms,
431 bool retransmission, 431 bool retransmission) {
432 int probe_cluster_id) {
433 if (SendingMedia() && ssrc == rtp_sender_.SSRC()) { 432 if (SendingMedia() && ssrc == rtp_sender_.SSRC()) {
434 return rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms, 433 return rtp_sender_.TimeToSendPacket(
435 retransmission, probe_cluster_id); 434 sequence_number, capture_time_ms, retransmission);
436 } 435 }
437 // No RTP sender is interested in sending this packet. 436 // No RTP sender is interested in sending this packet.
438 return true; 437 return true;
439 } 438 }
440 439
441 size_t ModuleRtpRtcpImpl::TimeToSendPadding(size_t bytes, 440 size_t ModuleRtpRtcpImpl::TimeToSendPadding(size_t bytes) {
442 int probe_cluster_id) { 441 return rtp_sender_.TimeToSendPadding(bytes);
443 return rtp_sender_.TimeToSendPadding(bytes, probe_cluster_id);
444 } 442 }
445 443
446 uint16_t ModuleRtpRtcpImpl::MaxPayloadLength() const { 444 uint16_t ModuleRtpRtcpImpl::MaxPayloadLength() const {
447 return rtp_sender_.MaxPayloadLength(); 445 return rtp_sender_.MaxPayloadLength();
448 } 446 }
449 447
450 uint16_t ModuleRtpRtcpImpl::MaxDataPayloadLength() const { 448 uint16_t ModuleRtpRtcpImpl::MaxDataPayloadLength() const {
451 return rtp_sender_.MaxDataPayloadLength(); 449 return rtp_sender_.MaxDataPayloadLength();
452 } 450 }
453 451
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989 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 987 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
990 StreamDataCountersCallback* callback) { 988 StreamDataCountersCallback* callback) {
991 rtp_sender_.RegisterRtpStatisticsCallback(callback); 989 rtp_sender_.RegisterRtpStatisticsCallback(callback);
992 } 990 }
993 991
994 StreamDataCountersCallback* 992 StreamDataCountersCallback*
995 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 993 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
996 return rtp_sender_.GetRtpStatisticsCallback(); 994 return rtp_sender_.GetRtpStatisticsCallback();
997 } 995 }
998 } // namespace webrtc 996 } // namespace webrtc
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