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Side by Side Diff: media/cast/net/cast_transport_sender_impl.cc

Issue 1837993002: Add log to detect the out of order RTCP packets. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 8 months ago
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1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "media/cast/net/cast_transport_sender_impl.h" 5 #include "media/cast/net/cast_transport_sender_impl.h"
6 6
7 #include <stddef.h> 7 #include <stddef.h>
8 #include <algorithm> 8 #include <algorithm>
9 #include <string> 9 #include <string>
10 #include <utility> 10 #include <utility>
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367 } else if (video_sender_ && video_sender_->ssrc() == ssrc) { 367 } else if (video_sender_ && video_sender_->ssrc() == ssrc) {
368 dedup_info.resend_interval = video_rtcp_session_->current_round_trip_time(); 368 dedup_info.resend_interval = video_rtcp_session_->current_round_trip_time();
369 369
370 // Only use audio stream to dedup if there is one. 370 // Only use audio stream to dedup if there is one.
371 if (audio_sender_) { 371 if (audio_sender_) {
372 dedup_info.last_byte_acked_for_audio = last_byte_acked_for_audio_; 372 dedup_info.last_byte_acked_for_audio = last_byte_acked_for_audio_;
373 } 373 }
374 } 374 }
375 375
376 if (!cast_message.missing_frames_and_packets.empty()) { 376 if (!cast_message.missing_frames_and_packets.empty()) {
377 // This call does two things. 377 VLOG(2) << "feedback_count: " << cast_message.feedback_count;
378 // 1. Specifies that retransmissions for packets not listed in the set are 378 // This call specifies a deduplication window. For video this would be the
379 // cancelled. 379 // most recent RTT. For audio there is no deduplication.
380 // 2. Specifies a deduplication window. For video this would be the most 380 ResendPackets(ssrc, cast_message.missing_frames_and_packets, false,
miu 2016/03/29 19:14:14 Looks like all possible callers of ResendPackets w
xjz 2016/03/29 19:33:49 As discussed, this is not applicable any more.
381 // recent RTT. For audio there is no deduplication.
382 ResendPackets(ssrc, cast_message.missing_frames_and_packets, true,
383 dedup_info); 381 dedup_info);
384 } 382 }
385 383
386 if (!cast_message.received_later_frames.empty()) { 384 if (!cast_message.received_later_frames.empty()) {
387 // Cancel resending frames that were received by the RTP receiver. 385 // Cancel resending frames that were received by the RTP receiver.
388 CancelSendingFrames(ssrc, cast_message.received_later_frames); 386 CancelSendingFrames(ssrc, cast_message.received_later_frames);
389 } 387 }
390 } 388 }
391 389
392 void CastTransportSenderImpl::AddValidRtpReceiver(uint32_t rtp_sender_ssrc, 390 void CastTransportSenderImpl::AddValidRtpReceiver(uint32_t rtp_sender_ssrc,
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486 "calling CastTransportSenderImpl::SendRtcpFromRtpReceiver."; 484 "calling CastTransportSenderImpl::SendRtcpFromRtpReceiver.";
487 return; 485 return;
488 } 486 }
489 pacer_.SendRtcpPacket(rtcp_builder_at_rtp_receiver_->local_ssrc(), 487 pacer_.SendRtcpPacket(rtcp_builder_at_rtp_receiver_->local_ssrc(),
490 rtcp_builder_at_rtp_receiver_->Finish()); 488 rtcp_builder_at_rtp_receiver_->Finish());
491 rtcp_builder_at_rtp_receiver_.reset(); 489 rtcp_builder_at_rtp_receiver_.reset();
492 } 490 }
493 491
494 } // namespace cast 492 } // namespace cast
495 } // namespace media 493 } // namespace media
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